Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined
Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets
I am looking for a SNOM phone guru that can create a dial plan for me with the following...strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765
i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [login to view URL] or yate or besip [login to view URL]
...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through SIP...
Looking to buy ready to use Call Centre CRM. Get back to me with demo. Complete documented...TAPI - Dialling from Windows applications • Text-to-Speech • VoIP Gateways • Voicemail • Visual Indicator for Message Waiting • Voicemail to email • Web Voicemail Interface Mobile data collection to push data into CRM Create reports on the fly.
I need Linphone rebranded with our Company info, logo, colors and only allow the setup of SIP accounts over TLS. Removal of options to create a linphone account. This must be done on: IOS, Android, Windows and OSX The result must be packages ready to deploy through app stores. and source code must be handed over to us, as Linphone is open source
...without ringing the phone. App will need to have standard features such as dashboards to see send data, user data, cost from carrier data. Must be able to connect to any SIP provider. Ringless Voicemail examples [login to view URL] [login to view URL] [login to view URL] https://www
Looking for a Tech Blogger to cover all the topics around VoIP, SIP trunking, DIDs, different countries legislations and Etc. Candidate needs to be familiar with the industry and love tech-savvy content Please when apply, send your samples
We need to bridge standard SIP calls to/from our iOs/Android app written in Adobe AIR Actionscript. In other words: handle the RTP media part of SIP to/from spk/mic. If you don't know by now what is required, PLEASE DO NOT RESPOND! Fixed payment $200.
I am unable to get calls from PSTN to freeswitch working. Calls from a SIP user into the switch (over the gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with
I need a SIP client COMPLETELY written in ActionScript, so NO external libraries or other dependencies. It should be able to connect with a SIP server, ACCEPT calls only (so don't worry about dialing and invites) and handle that 2-way phone call (mic/speaker). That's it, nothing specific! If you know what you're doing, you don't need anything else
I require a voicemail d...csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [login to view URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.
we are looking for en expert to develop a app that will act as sip gsm gateway i am including here a [login to view URL] to a software that was develop for window [login to view URL] 2, the actual software for window s i will drop it to you once we hire you 3. a suggestion
Build a front-end with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar ...with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar PHP framework, manipulate the Asterisk PBX from the interface (ivr, Sip/iax, did, ami/agi, voicemail, routes etc.)
...SMS Gateway service that utilizes either PBX, SMPP, or VoIP - the platform must be able to run stand-alone meaning it can not use any 3rd party services such as a 3rd party sip provider, a 3rd party SMS service such as Twilio or Plivo. It must not need SIM Cards or modems. It must be able to create VoIP numbers (on it's own) ex; not using Google Voice
...v2/v3/v5/ i want to run 32calls you can connect server to local a VPN and MASQUERADE connections or you can install any SIP PBX on local route like sofia-sip free-switch and register server to local if you want you can install a SIP Signal service on openwrt and pass call to voip device also you can bridge network and assin a server IP on local Voip
...expert to fulfill the following: 1- Write a script for VICIDIAL installation with a single command line from [login to view URL] 2- Tutor the proper basic setup for interconnecting with SIP trunk 3- Tutor on how to use VICIDIAL in the following concepts: a- Press-1 campaign: where we will send a pre-recorded message that has an offer, if the customer is interested
Need a browser extension or application for both windows and mac. The soft phone will require an agent to log in and be able to show when an ...from one agent to another agent. All call details will need to captured - Date & Time, Caller ID (Number) Start of Call and Finish, Agent ID Needs to be able to work across SIP and PBX providers using TAPI
...interface to create extensions, reports, online and offline ext, and PBX functions FUNCTIONS: - queues; - reports; (with export to .pdf and .csv file) - IVR; - group capture ext; - SIP TRUNK configuration; - DAC; Mandatory: - send documentation from development; - log from customers and admins about changes; - each user have your color/logo (size from picture
Phase 1 Need to setup asteriskNow with ippbx (tata telecom). Setting up sip trunk. redirecting calls on specific numbers to specific sip no. At specific work hours or if specified through API, then to redirect calls to Mobile no. Setup recording of calls.
...using Zoiper as softphone application to register extensions with IPPBX System. Our employees using Zoiper from inside and outside office. The protocols we are using are SIP and IAX We started a new office in Egypt, and would like to use Zoiper to register with same IPPBX System by employees working there. Unfortunately, its not working. We have
I need the Linphone client rebranded for Android/IOS and Windows/OSX. Simply with our logo / app-icon, removing options for creating or using linphone account, so only sip account is configurable. Furthermore changed to our colors. Packages for IOS and Android should be ready for deployment to the public stores, so we need the sourcecode when
Datalink International is seeking a skilled Android and iOS Application developer. Must have 5 + years of proven development skills. All A...developer. Must have 5 + years of proven development skills. All Apps developed will link over GSM Cellular networks to existing GPS and SMS Host Software. Must be familiar with SIP and serial data connections
...that allow me to send calls to the mobile via SIP ( VoIP ) and this calls can me dialed ( Outbound ) Via the GSM ( the SIM card which is installed in the phone ) In more details Now we have a mobile device and Asterisk server, I want to link the mobile to the server and this server will send the call (SIP) and the mobile will make this call via the SIM
I want an Android which has a few calculators - Loan EMI Fixed Deposits Recurring Deposits Direct MF / SIP Simple Interest Compound Interest Target version must be Android 8 Oreo and minimum supported version should be Lollipop. Design of app should be good.
...create everyday a new CDR Table i need a CDR page where i can check all the calls report like cdr by account name/gateway name or ip/client name or ip/ make sure i can use a SIP account as a routing gateway I want to make same to same VOS3000 softswitch If you want to Use asterisk Use JAVA-AGI database must be PostgreSQL Priority musbe be have to work
...overlay, once we select "icon overlay" option in right click menu on file or folder. we need it in a way without affecting security/safety of the macOS and without we disable SIP. the app will sync file/folder contents, so a syncing file has different icon than a sync'd file. please look at the link below for more information about what we need to do
Hey! I'm looking for someone knows how to work with TwiML Voice API, I would like to create a Queue with Wait Music, and each caller getting numbe...to work with TwiML Voice API, I would like to create a Queue with Wait Music, and each caller getting number Until 1 and then redirected to agent. * Agents getting calls by SIP address. * Works remotely.
...solution based on DNS SRV records. Need somebody to provide me a proper configuration tutorial and example of usage. My scenario is: Have two SIP servers in two different locations. Both running Vodia SIP PBX. Vodia using multi tenant subdomain structure based on sub-domains. Each tenant is a sub-domain. DNS provider is Godaddy and we have DNS plus
Hi, I need to get native apps for android and ios for SIP Audio calling, Video not needed. Calling can be done app to app and on phone number as well. SIP Gateway would be provided. You have to manage for app to app call Something like rebtel. Customer can buy the package from app/website and call via website/app There should be front end, super
...codes that exist) On phone 1 You will create a app which accepts SIP calls and generates dtmf or voice dial commands based on what comes in the sip header. the sip app will register to a sip sever who will send it numbers to be dialed in the header When the sip server sends a call, the sip app will issue dial command via the headphone speaker. (to pho...
...Project. The current state is pretty simple and straight forward. Just a PBX with 4 extensions, and a trunk. All works well. This is shown in the diagram as 'Before'. One SIP phone however is very special. It doesn't have a transfer button. So no call transfer can be initiated on the phone. It should be possible however to initiate a transfer via
...initially provide reporting capabilities on Twilio's SIP trunking product. Using Twilio's REST API, you'll build the following: 1. User auth 2. Dashboard with usage data (much like what twilio provides now, but for the subaccount) 3. Historical reporting - whatever twilio provides data-wise on the SIP trunk side of the house (call date/time, ANI, duration
Hi. The consultant will remotely help us finish the setup of IP pbx elastix. Make sure that all calls are secure...fail2ban etc..ensure a sip trace log on demand with setup of a function...
i have a freepbx system and need to configure one sip detail but not able to use that details like as sip trunk. if i put same details on xlite or IP phone its working fine. so its a simple task but i need some expert one for this.
i got this error in asterisk and freepbx under centos "The number you dialed is not in service, please check your number and try again" im using twilio sip trunk, please i need soeone who fixed, everything i twilio is already done
...CarrierBid. I have 40 more icons that I need designed for the following subjects: 1. Free WiFi 2. WiFi Router 3. Network Access Point 4. Customers walking 5. Just the word: “SIP” with a telephone handset 6. Paper icon with the words: RFP RFQ RFI with a $ symbol 7. Language Translation services 8. iphone notification 9. Increased sales 10. Advertising
Hey, guys, I have a simple project. I have a FreePBX server that I need to build an inbound Sip trunk to 2 separate carriers + build inbound routes for them. It should be a simple process. Please respond to the bid with "What up Dingo" at the beginning of your message so that i know you have read.